# coding=utf-8 # Copyright 2021 The TensorFlow Datasets Authors and the HuggingFace Datasets Authors. # # Licensed under the Apache License, Version 2.0 (the "License"); # you may not use this file except in compliance with the License. # You may obtain a copy of the License at # # http://www.apache.org/licenses/LICENSE-2.0 # # Unless required by applicable law or agreed to in writing, software # distributed under the License is distributed on an "AS IS" BASIS, # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. # See the License for the specific language governing permissions and # limitations under the License. # Lint as: python3 """SUPERB: Speech processing Universal PERformance Benchmark.""" import csv import glob import os import textwrap from dataclasses import dataclass import datasets _CITATION = """\ @article{DBLP:journals/corr/abs-2105-01051, author = {Shu{-}Wen Yang and Po{-}Han Chi and Yung{-}Sung Chuang and Cheng{-}I Jeff Lai and Kushal Lakhotia and Yist Y. Lin and Andy T. Liu and Jiatong Shi and Xuankai Chang and Guan{-}Ting Lin and Tzu{-}Hsien Huang and Wei{-}Cheng Tseng and Ko{-}tik Lee and Da{-}Rong Liu and Zili Huang and Shuyan Dong and Shang{-}Wen Li and Shinji Watanabe and Abdelrahman Mohamed and Hung{-}yi Lee}, title = {{SUPERB:} Speech processing Universal PERformance Benchmark}, journal = {CoRR}, volume = {abs/2105.01051}, year = {2021}, url = {https://arxiv.org/abs/2105.01051}, archivePrefix = {arXiv}, eprint = {2105.01051}, timestamp = {Thu, 01 Jul 2021 13:30:22 +0200}, biburl = {https://dblp.org/rec/journals/corr/abs-2105-01051.bib}, bibsource = {dblp computer science bibliography, https://dblp.org} } """ _DESCRIPTION = """\ Self-supervised learning (SSL) has proven vital for advancing research in natural language processing (NLP) and computer vision (CV). The paradigm pretrains a shared model on large volumes of unlabeled data and achieves state-of-the-art (SOTA) for various tasks with minimal adaptation. However, the speech processing community lacks a similar setup to systematically explore the paradigm. To bridge this gap, we introduce Speech processing Universal PERformance Benchmark (SUPERB). SUPERB is a leaderboard to benchmark the performance of a shared model across a wide range of speech processing tasks with minimal architecture changes and labeled data. Among multiple usages of the shared model, we especially focus on extracting the representation learned from SSL due to its preferable re-usability. We present a simple framework to solve SUPERB tasks by learning task-specialized lightweight prediction heads on top of the frozen shared model. Our results demonstrate that the framework is promising as SSL representations show competitive generalizability and accessibility across SUPERB tasks. We release SUPERB as a challenge with a leaderboard and a benchmark toolkit to fuel the research in representation learning and general speech processing. Note that in order to limit the required storage for preparing this dataset, the audio is stored in the .wav format and is not converted to a float32 array. To convert the audio file to a float32 array, please make use of the `.map()` function as follows: ```python import soundfile as sf def map_to_array(batch): speech_array, _ = sf.read(batch["file"]) batch["speech"] = speech_array return batch dataset = dataset.map(map_to_array, remove_columns=["file"]) ``` """ class SuperbConfig(datasets.BuilderConfig): """BuilderConfig for Superb.""" def __init__( self, features, url, data_url=None, supervised_keys=None, **kwargs, ): super().__init__(version=datasets.Version("1.9.0", ""), **kwargs) self.features = features self.data_url = data_url self.url = url self.supervised_keys = supervised_keys class Superb(datasets.GeneratorBasedBuilder): """Superb dataset.""" BUILDER_CONFIGS = [ SuperbConfig( name="asr", description=textwrap.dedent( """\ ASR transcribes utterances into words. While PR analyzes the improvement in modeling phonetics, ASR reflects the significance of the improvement in a real-world scenario. LibriSpeech train-clean-100/dev-clean/test-clean subsets are used for training/validation/testing. The evaluation metric is word error rate (WER).""" ), features=datasets.Features( { "file": datasets.Value("string"), "audio": datasets.features.Audio(sampling_rate=16_000), "text": datasets.Value("string"), "speaker_id": datasets.Value("int64"), "chapter_id": datasets.Value("int64"), "id": datasets.Value("string"), } ), supervised_keys=("file", "text"), url="http://www.openslr.org/12", data_url="http://www.openslr.org/resources/12/", ), SuperbConfig( name="ks", description=textwrap.dedent( """\ Keyword Spotting (KS) detects preregistered keywords by classifying utterances into a predefined set of words. The task is usually performed on-device for the fast response time. Thus, accuracy, model size, and inference time are all crucial. SUPERB uses the widely used Speech Commands dataset v1.0 for the task. The dataset consists of ten classes of keywords, a class for silence, and an unknown class to include the false positive. The evaluation metric is accuracy (ACC)""" ), features=datasets.Features( { "file": datasets.Value("string"), "audio": datasets.features.Audio(sampling_rate=16_000), "label": datasets.ClassLabel( names=[ "yes", "no", "up", "down", "left", "right", "on", "off", "stop", "go", "_silence_", "_unknown_", ] ), } ), supervised_keys=("file", "label"), url="https://www.tensorflow.org/datasets/catalog/speech_commands", data_url="http://download.tensorflow.org/data/{filename}", ), SuperbConfig( name="ic", description=textwrap.dedent( """\ Intent Classification (IC) classifies utterances into predefined classes to determine the intent of speakers. SUPERB uses the Fluent Speech Commands dataset, where each utterance is tagged with three intent labels: action, object, and location. The evaluation metric is accuracy (ACC).""" ), features=datasets.Features( { "file": datasets.Value("string"), "audio": datasets.features.Audio(sampling_rate=16_000), "speaker_id": datasets.Value("string"), "text": datasets.Value("string"), "action": datasets.ClassLabel( names=["activate", "bring", "change language", "deactivate", "decrease", "increase"] ), "object": datasets.ClassLabel( names=[ "Chinese", "English", "German", "Korean", "heat", "juice", "lamp", "lights", "music", "newspaper", "none", "shoes", "socks", "volume", ] ), "location": datasets.ClassLabel(names=["bedroom", "kitchen", "none", "washroom"]), } ), supervised_keys=None, url="https://fluent.ai/fluent-speech-commands-a-dataset-for-spoken-language-understanding-research/", data_url="http://fluent.ai:2052/jf8398hf30f0381738rucj3828chfdnchs.tar.gz", ), SuperbConfig( name="si", description=textwrap.dedent( """\ Speaker Identification (SI) classifies each utterance for its speaker identity as a multi-class classification, where speakers are in the same predefined set for both training and testing. The widely used VoxCeleb1 dataset is adopted, and the evaluation metric is accuracy (ACC).""" ), features=datasets.Features( { "file": datasets.Value("string"), "audio": datasets.features.Audio(sampling_rate=16_000), # VoxCeleb1 contains 1251 speaker IDs in range ["id10001",..."id11251"] "label": datasets.ClassLabel(names=[f"id{i + 10001}" for i in range(1251)]), } ), supervised_keys=("file", "label"), url="https://www.robots.ox.ac.uk/~vgg/data/voxceleb/vox1.html", ), SuperbConfig( name="sd", description=textwrap.dedent( """\ Speaker Diarization (SD) predicts `who is speaking when` for each timestamp, and multiple speakers can speak simultaneously. The model has to encode rich speaker characteristics for each frame and should be able to represent mixtures of signals. [LibriMix] is adopted where LibriSpeech train-clean-100/dev-clean/test-clean are used to generate mixtures for training/validation/testing. We focus on the two-speaker scenario as the first step. The time-coded speaker labels were generated using alignments from Kaldi LibriSpeech ASR model. The evaluation metric is diarization error rate (DER).""" ), features=datasets.Features( { "record_id": datasets.Value("string"), "file": datasets.Value("string"), "audio": datasets.features.Audio(sampling_rate=16_000), "start": datasets.Value("int64"), "end": datasets.Value("int64"), "speakers": [ { "speaker_id": datasets.Value("string"), "start": datasets.Value("int64"), "end": datasets.Value("int64"), } ], } ), # TODO supervised_keys=None, # TODO url="https://github.com/ftshijt/LibriMix", data_url="https://huggingface.co./datasets/superb/superb-data/resolve/main/sd/{split}/{filename}", ), SuperbConfig( name="er", description=textwrap.dedent( """\ Emotion Recognition (ER) predicts an emotion class for each utterance. The most widely used ER dataset IEMOCAP is adopted, and we follow the conventional evaluation protocol: we drop the unbalanced emotion classes to leave the final four classes with a similar amount of data points and cross-validate on five folds of the standard splits. The evaluation metric is accuracy (ACC).""" ), features=datasets.Features( { "file": datasets.Value("string"), "audio": datasets.features.Audio(sampling_rate=16_000), "label": datasets.ClassLabel(names=["neu", "hap", "ang", "sad"]), } ), supervised_keys=("file", "label"), url="https://sail.usc.edu/iemocap/", ), ] @property def manual_download_instructions(self): if self.config.name == "si": return textwrap.dedent( """ Please download the VoxCeleb dataset using the following script, which should create `VoxCeleb1/wav/id*` directories for both train and test speakers`: ``` mkdir VoxCeleb1 cd VoxCeleb1 wget https://thor.robots.ox.ac.uk/~vgg/data/voxceleb/vox1a/vox1_dev_wav_partaa wget https://thor.robots.ox.ac.uk/~vgg/data/voxceleb/vox1a/vox1_dev_wav_partab wget https://thor.robots.ox.ac.uk/~vgg/data/voxceleb/vox1a/vox1_dev_wav_partac wget https://thor.robots.ox.ac.uk/~vgg/data/voxceleb/vox1a/vox1_dev_wav_partad cat vox1_dev* > vox1_dev_wav.zip unzip vox1_dev_wav.zip wget https://thor.robots.ox.ac.uk/~vgg/data/voxceleb/vox1a/vox1_test_wav.zip unzip vox1_test_wav.zip # download the official SUPERB train-dev-test split wget https://raw.githubusercontent.com/s3prl/s3prl/master/s3prl/downstream/voxceleb1/veri_test_class.txt ```""" ) elif self.config.name == "er": return textwrap.dedent( """ Please download the IEMOCAP dataset after submitting the request form here: https://sail.usc.edu/iemocap/iemocap_release.htm Having downloaded the dataset you can extract it with `tar -xvzf IEMOCAP_full_release.tar.gz` which should create a folder called `IEMOCAP_full_release` """ ) return None def _info(self): return datasets.DatasetInfo( description=_DESCRIPTION, features=self.config.features, supervised_keys=self.config.supervised_keys, homepage=self.config.url, citation=_CITATION, ) def _split_generators(self, dl_manager): if self.config.name == "asr": _DL_URLS = { "dev": self.config.data_url + "dev-clean.tar.gz", "test": self.config.data_url + "test-clean.tar.gz", "train": self.config.data_url + "train-clean-100.tar.gz", } archive_path = dl_manager.download_and_extract(_DL_URLS) return [ datasets.SplitGenerator(name=datasets.Split.TRAIN, gen_kwargs={"archive_path": archive_path["train"]}), datasets.SplitGenerator( name=datasets.Split.VALIDATION, gen_kwargs={"archive_path": archive_path["dev"]} ), datasets.SplitGenerator(name=datasets.Split.TEST, gen_kwargs={"archive_path": archive_path["test"]}), ] elif self.config.name == "ks": _DL_URLS = { "train_val_test": self.config.data_url.format(filename="speech_commands_v0.01.tar.gz"), "test": self.config.data_url.format(filename="speech_commands_test_set_v0.01.tar.gz"), } archive_path = dl_manager.download_and_extract(_DL_URLS) return [ datasets.SplitGenerator( name=datasets.Split.TRAIN, gen_kwargs={"archive_path": archive_path["train_val_test"], "split": "train"}, ), datasets.SplitGenerator( name=datasets.Split.VALIDATION, gen_kwargs={"archive_path": archive_path["train_val_test"], "split": "val"}, ), datasets.SplitGenerator( name=datasets.Split.TEST, gen_kwargs={"archive_path": archive_path["test"], "split": "test"} ), ] elif self.config.name == "ic": archive_path = dl_manager.download_and_extract(self.config.data_url) return [ datasets.SplitGenerator( name=datasets.Split.TRAIN, gen_kwargs={"archive_path": archive_path, "split": "train"}, ), datasets.SplitGenerator( name=datasets.Split.VALIDATION, gen_kwargs={"archive_path": archive_path, "split": "valid"}, ), datasets.SplitGenerator( name=datasets.Split.TEST, gen_kwargs={"archive_path": archive_path, "split": "test"} ), ] elif self.config.name == "si": manual_dir = os.path.abspath(os.path.expanduser(dl_manager.manual_dir)) return [ datasets.SplitGenerator( name=datasets.Split.TRAIN, gen_kwargs={"archive_path": manual_dir, "split": 1}, ), datasets.SplitGenerator( name=datasets.Split.VALIDATION, gen_kwargs={"archive_path": manual_dir, "split": 2}, ), datasets.SplitGenerator(name=datasets.Split.TEST, gen_kwargs={"archive_path": manual_dir, "split": 3}), ] elif self.config.name == "sd": splits = ["train", "dev", "test"] _DL_URLS = { split: { filename: self.config.data_url.format(split=split, filename=filename) for filename in ["reco2dur", "segments", "utt2spk", "wav.zip"] } for split in splits } archive_path = dl_manager.download_and_extract(_DL_URLS) return [ datasets.SplitGenerator( name=datasets.NamedSplit(split), gen_kwargs={"archive_path": archive_path[split], "split": split} ) for split in splits ] elif self.config.name == "er": manual_dir = os.path.abspath(os.path.expanduser(dl_manager.manual_dir)) return [ datasets.SplitGenerator( name=f"session{i}", gen_kwargs={"archive_path": manual_dir, "split": i}, ) for i in range(1, 6) ] def _generate_examples(self, archive_path, split=None): """Generate examples.""" if self.config.name == "asr": transcripts_glob = os.path.join(archive_path, "LibriSpeech", "*", "*", "*", "*.txt") key = 0 for transcript_path in sorted(glob.glob(transcripts_glob)): transcript_dir_path = os.path.dirname(transcript_path) with open(transcript_path, "r", encoding="utf-8") as f: for line in f: line = line.strip() id_, transcript = line.split(" ", 1) audio_file = f"{id_}.flac" speaker_id, chapter_id = [int(el) for el in id_.split("-")[:2]] audio_path = os.path.join(transcript_dir_path, audio_file) yield key, { "id": id_, "speaker_id": speaker_id, "chapter_id": chapter_id, "file": audio_path, "audio": audio_path, "text": transcript, } key += 1 elif self.config.name == "ks": words = ["yes", "no", "up", "down", "left", "right", "on", "off", "stop", "go"] splits = _split_ks_files(archive_path, split) for key, audio_file in enumerate(sorted(splits[split])): base_dir, file_name = os.path.split(audio_file) _, word = os.path.split(base_dir) if word in words: label = word elif word == "_silence_" or word == "_background_noise_": label = "_silence_" else: label = "_unknown_" yield key, {"file": audio_file, "audio": audio_file, "label": label} elif self.config.name == "ic": root_path = os.path.join(archive_path, "fluent_speech_commands_dataset") csv_path = os.path.join(root_path, "data", f"{split}_data.csv") with open(csv_path, encoding="utf-8") as csv_file: csv_reader = csv.reader(csv_file, delimiter=",", skipinitialspace=True) next(csv_reader) for row in csv_reader: key, file_path, speaker_id, text, action, object_, location = row audio_path = os.path.join(root_path, file_path) yield key, { "file": audio_path, "audio": audio_path, "speaker_id": speaker_id, "text": text, "action": action, "object": object_, "location": location, } elif self.config.name == "si": wav_path = os.path.join(archive_path, "wav") splits_path = os.path.join(archive_path, "veri_test_class.txt") with open(splits_path, "r", encoding="utf-8") as f: for key, line in enumerate(f): split_id, file_path = line.strip().split(" ") if int(split_id) != split: continue speaker_id = file_path.split("/")[0] audio_path = os.path.join(wav_path, file_path) yield key, { "file": audio_path, "audio": audio_path, "label": speaker_id, } elif self.config.name == "sd": data = SdData(archive_path) args = SdArgs() chunk_indices = _generate_chunk_indices(data, args, split=split) if split != "test": for key, (rec, st, ed) in enumerate(chunk_indices): speakers = _get_speakers(rec, data, args) yield key, { "record_id": rec, "file": data.wavs[rec], "audio": data.wavs[rec], "start": st, "end": ed, "speakers": speakers, } else: key = 0 for rec in chunk_indices: for rec, st, ed in chunk_indices[rec]: speakers = _get_speakers(rec, data, args) yield key, { "record_id": rec, "file": data.wavs[rec], "audio": data.wavs[rec], "start": st, "end": ed, "speakers": speakers, } key += 1 elif self.config.name == "er": root_path = os.path.join(archive_path, f"Session{split}") wav_path = os.path.join(root_path, "sentences", "wav") labels_path = os.path.join(root_path, "dialog", "EmoEvaluation", "*.txt") emotions = ["neu", "hap", "ang", "sad", "exc"] key = 0 for labels_file in sorted(glob.glob(labels_path)): with open(labels_file, "r", encoding="utf-8") as f: for line in f: if line[0] != "[": continue _, filename, emo, _ = line.split("\t") if emo not in emotions: continue wav_subdir = filename.rsplit("_", 1)[0] filename = f"{filename}.wav" audio_path = os.path.join(wav_path, wav_subdir, filename) yield key, { "file": audio_path, "audio": audio_path, "label": emo.replace("exc", "hap"), } key += 1 class SdData: def __init__(self, data_dir): """Load sd data.""" self.segments = self._load_segments_rechash(data_dir["segments"]) self.utt2spk = self._load_utt2spk(data_dir["utt2spk"]) self.wavs = self._load_wav_zip(data_dir["wav.zip"]) self.reco2dur = self._load_reco2dur(data_dir["reco2dur"]) def _load_segments_rechash(self, segments_file): """Load segments file as dict with recid index.""" ret = {} if not os.path.exists(segments_file): return None with open(segments_file, encoding="utf-8") as f: for line in f: utt, rec, st, et = line.strip().split() if rec not in ret: ret[rec] = [] ret[rec].append({"utt": utt, "st": float(st), "et": float(et)}) return ret def _load_wav_zip(self, wav_zip): """Return dictionary { rec: wav_rxfilename }.""" wav_dir = os.path.join(wav_zip, "wav") return { os.path.splitext(filename)[0]: os.path.join(wav_dir, filename) for filename in sorted(os.listdir(wav_dir)) } def _load_utt2spk(self, utt2spk_file): """Returns dictionary { uttid: spkid }.""" with open(utt2spk_file, encoding="utf-8") as f: lines = [line.strip().split(None, 1) for line in f] return {x[0]: x[1] for x in lines} def _load_reco2dur(self, reco2dur_file): """Returns dictionary { recid: duration }.""" if not os.path.exists(reco2dur_file): return None with open(reco2dur_file, encoding="utf-8") as f: lines = [line.strip().split(None, 1) for line in f] return {x[0]: float(x[1]) for x in lines} @dataclass class SdArgs: chunk_size: int = 2000 frame_shift: int = 160 subsampling: int = 1 label_delay: int = 0 num_speakers: int = 2 rate: int = 16000 use_last_samples: bool = True def _generate_chunk_indices(data, args, split=None): chunk_indices = [] if split != "test" else {} # make chunk indices: filepath, start_frame, end_frame for rec in data.wavs: data_len = int(data.reco2dur[rec] * args.rate / args.frame_shift) data_len = int(data_len / args.subsampling) if split == "test": chunk_indices[rec] = [] if split != "test": for st, ed in _gen_frame_indices( data_len, args.chunk_size, args.chunk_size, args.use_last_samples, label_delay=args.label_delay, subsampling=args.subsampling, ): chunk_indices.append((rec, st * args.subsampling, ed * args.subsampling)) else: for st, ed in _gen_chunk_indices(data_len, args.chunk_size): chunk_indices[rec].append((rec, st * args.subsampling, ed * args.subsampling)) return chunk_indices def _count_frames(data_len, size, step): # no padding at edges, last remaining samples are ignored return int((data_len - size + step) / step) def _gen_frame_indices(data_length, size=2000, step=2000, use_last_samples=False, label_delay=0, subsampling=1): i = -1 for i in range(_count_frames(data_length, size, step)): yield i * step, i * step + size if use_last_samples and i * step + size < data_length: if data_length - (i + 1) * step - subsampling * label_delay > 0: yield (i + 1) * step, data_length def _gen_chunk_indices(data_len, chunk_size): step = chunk_size start = 0 while start < data_len: end = min(data_len, start + chunk_size) yield start, end start += step def _get_speakers(rec, data, args): return [ { "speaker_id": data.utt2spk[segment["utt"]], "start": round(segment["st"] * args.rate / args.frame_shift), "end": round(segment["et"] * args.rate / args.frame_shift), } for segment in data.segments[rec] ] def _split_ks_files(archive_path, split): audio_path = os.path.join(archive_path, "**", "*.wav") audio_paths = glob.glob(audio_path) if split == "test": # use all available files for the test archive return {"test": audio_paths} val_list_file = os.path.join(archive_path, "validation_list.txt") test_list_file = os.path.join(archive_path, "testing_list.txt") with open(val_list_file, encoding="utf-8") as f: val_paths = f.read().strip().splitlines() val_paths = [os.path.join(archive_path, p) for p in val_paths] with open(test_list_file, encoding="utf-8") as f: test_paths = f.read().strip().splitlines() test_paths = [os.path.join(archive_path, p) for p in test_paths] # the paths for the train set is just whichever paths that do not exist in # either the test or validation splits train_paths = list(set(audio_paths) - set(val_paths) - set(test_paths)) return {"train": train_paths, "val": val_paths}